From the log, it showed lookup(): 'account name' Not found in usrloc.
I think it is the NAT problem, so I use stun for the case below.
Telephone (A) | PSTN | G/W | openser | NAT -- IP phone(B) -- STUN + ----- IP phone(C) ------+
Below is the result: A to B/C is ok B/C to A is ok However, there is no sound when B to C or vice versa. What reason will cause no sound between B and C? Is the the reason from the NAT/STUN?
On 12/12/05, Klaus Darilion klaus.mailinglists@pernau.at wrote:
Use ngrep to watch for incoming SIP requests on the SIP proxy.
Take a look at the logfiles on the gateway.
klaus
unplug wrote:
Below is the common configuration of the network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to make call from Telephone to softphone. I wonder why it happened and any reference to trace the problem. Anyone have such experience?
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